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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2ae140ae7e34bd1e102ca832a14b7b0adbb72d15
/
pc
/
rtcstatscollector.cc
57858b3
Reland "Update RTCStatsCollector to work with RtpTransceivers"
by Steve Anton
· 7 years ago
ee2388f
Revert "Update RTCStatsCollector to work with RtpTransceivers"
by Guido Urdaneta
· 7 years ago
56bae8d
Update RTCStatsCollector to work with RtpTransceivers
by Steve Anton
· 7 years ago
5dfde18
Change PeerConnection stats interface to be more flexible
by Steve Anton
· 7 years ago
76d2952
Don't crash when sender info has been discarded by lower layers.
by Harald Alvestrand
· 7 years ago
2d8609c
Move internal PeerConnection methods to PeerConnectionInternal
by Steve Anton
· 7 years ago
b8e1201
Generate track stats when SSRC=0
by Harald Alvestrand
· 7 years ago
a3dab84
Refactor stream stats generation
by Harald Alvestrand
· 7 years ago
c72af93
Reland "Move stats ID generation from SSRC to local ID"
by Harald Alvestrand
· 7 years ago
c0092c3
Revert "Move stats ID generation from SSRC to local ID"
by Erik Språng
· 7 years ago
e357a4d
Move stats ID generation from SSRC to local ID
by Harald Alvestrand
· 7 years ago
8906187
Pivot generation of stats to iterate senders/receivers
by Harald Alvestrand
· 7 years ago
593e325
Change RTCStatsCollector to only access channels from signaling thread
by Steve Anton
· 7 years ago
719487e
Generate signed packets_lost in WebRTC-stats
by Harald Alvestrand
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
37e489c
Add network_type to local RTCIceCandidateStats
by Gary Liu
· 7 years ago
89e7126
Optional: Use nullopt and implicit construction in /pc/rtcstatscollector.cc
by Oskar Sundbom
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
d5585ca
Move almost all references from WebRtcSession to PeerConnection
by Steve Anton
· 7 years ago
e2d6a06
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
1af3d82
Revert "Reland "Clean up libjingle API dependencies.""
by Henrik Kjellander
· 7 years ago
9185aca
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
978b876
Move clients of WebRtcSession to use PeerConnection
by Steve Anton
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 7 years ago
581df61
Revert "Reland "Clean up libjingle API dependencies.""
by Patrik Höglund
· 7 years ago
5117b04
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
7bcfc3b
Revert "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
bf66794
Revert "Move clients of WebRtcSession to use PeerConnection"
by Alex Loiko
· 7 years ago
57fb315
Clean up libjingle API dependencies.
by Patrik Höglund
· 7 years ago
3dc4d4a
Move clients of WebRtcSession to use PeerConnection
by Steve Anton
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/pc/rtcstatscollector.cc]
8ab0fd8
Reland of Trace the stats report as JSON instead of each stat separately. (patchset #1 id:1 of https://codereview.webrtc.org/3001683002/ )
by ehmaldonado
· 7 years ago
2dbc69f
Add stats totalSamplesReceived and concealedSamples
by Steve Anton
· 7 years ago
3439c89
Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ )
by mbonadei
· 7 years ago
80c6578
Trace the stats report as JSON instead of each stat separately.
by ehmaldonado
· 7 years ago
80c829f
Enable tracing on rtcstats_integrationtest.cc
by ehmaldonado
· 7 years ago
b0b721a
Increase the size of the buffer for type.name.id.
by ehmaldonado
· 7 years ago
a26196b
Trace stats in RTCStatsCollector.
by ehmaldonado
· 7 years ago
e76bd3a
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
by zstein
· 7 years ago
7b0c6fa
RTCStatsCollector: Get track IDs from senders/receivers instead of streams.
by hbos
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
f79ade1
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
by stefan
· 7 years ago
d72098a
Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )
by charujain
· 7 years ago
e80f4c9
Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
by Stefan Holmer
· 7 years ago
5bf9def
RTCStatsCollector: Remove closed channels from opened set.
by hbos
· 7 years ago
a7a9be1
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
by hbos
· 8 years ago
13f54b2
Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine.
by hbos
· 8 years ago
bf8d3e5
RTCIceCandidatePairStats.[total/current]RoundTripTime collected.
by hbos
· 8 years ago
92eaec6
RTCIceCandidatePairStats.nominated collected.
by hbos
· 8 years ago
a51d4f3
Re-land of RTCInboundRTPStreamStats.qpSum collected.
by hbos
· 8 years ago
ed02c6d
Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ )
by skvlad
· 8 years ago
cd195be
RTCInboundRTPStreamStats.qpSum collected.
by hbos
· 8 years ago
338f78a
RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected.
by hbos
· 8 years ago
3443bb7
RTCRTPStreamStats.ssrc changed type to uint32_t.
by hbos
· 8 years ago
585a9b1
Refactor and clean-up relating to RTCCodecStats.
by hbos
· 8 years ago
e702b30
Adding C++ versions of currently spec'd "RtpParameters" structs.
by deadbeef
· 8 years ago
b0ae920
RTCRTPStreamStats.mediaTrackId renamed to trackId.
by hbos
· 8 years ago
50cfe1f
RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
by hbos
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (99%) from webrtc/api/rtcstatscollector.cc]
f64941f
RTCMediaStreamTrackStats.framesDecoded collected.
by hbos
· 8 years ago
fefe076
RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector.
by hbos
· 8 years ago
42f6d2f
RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector.
by hbos
· 8 years ago
9e30274
RTCMediaStreamTrackStats collected on a per-attachment basis.
by hbos
· 8 years ago
160e4a7
RTCMediaStreamTrackStats.kind added and collected.
by hbos
· 8 years ago
7064d59
RTCTransportStats.dtlsState replaces .activeConnection
by hbos
· 8 years ago
84abeb1
RTC[In/Out]boundRTPStreamStats.mediaTrackId collected.
by hbos
· 8 years ago
953c2ce
Reland of: Separating SCTP code from BaseChannel/MediaChannel.
by deadbeef
· 8 years ago
c0dad89
Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
by deadbeef
· 8 years ago
67b3bbe
Separating SCTP code from BaseChannel/MediaChannel.
by deadbeef
· 8 years ago
b4e426e
RTCIceCandidateStats.transportId added and collected.
by hbos
· 8 years ago
6769c49
RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
by hbos
· 8 years ago
06495bc
RTCIceCandidatePairStats.[state/priority] added, ConnectionInfo updated.
by hbos
· 8 years ago
f415f8a
Removed RTCStatsCollector::ProducePartialResultsOnWorkerThread.
by hbos
· 8 years ago
02d2a92
RTCStatsReport::AddStats DCHECKs that the ID is unique.
by hbos
· 8 years ago
b78306a
Fix segfault when PeerConnection is destroyed during stats collection.
by hbos
· 8 years ago
df6075a
RTCStatsCollector: Utilize network thread to minimize thread hops.
by hbos
· 8 years ago
3168c7a
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
by hbos
· 8 years ago
e10e6d1
RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.
by hbos
· 8 years ago
9a394f0
Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value.
by hbos
· 8 years ago
e448dd5
RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector
by hbos
· 8 years ago
02cd4d6
RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.
by hbos
· 8 years ago
d82f512
RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector.
by hbos
· 8 years ago
0583b28
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
by hbos
· 8 years ago
0adb828
RTCCodecStats[1] added.
by hbos
· 8 years ago
820f578
RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.
by hbos
· 8 years ago
82ebe02
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
by hbos
· 8 years ago
1b35d4c
Don't copy loop variables when we don't have to
by kwiberg
· 8 years ago
09bc128
RTCMediaStream[Track]Stats added.
by hbos
· 8 years ago
eeafe94
RTCInboundRTPStreamStats[1] added.
by hbos
· 8 years ago
6ded190
RTCOutboundRTPStreamStats[1] added.
by hbos
· 8 years ago
7eaa836
Revert of RTCOutboundRTPStreamStats added. (patchset #3 id:80001 of https://codereview.webrtc.org/2456463002/ )
by perkj
· 8 years ago
4ed0750
Revert of RTCInboundRTPStreamStats added. (patchset #4 id:100001 of https://codereview.webrtc.org/2452043002/ )
by perkj
· 8 years ago
0d7bf16
RTCInboundRTPStreamStats[1] added.
by hbos
· 8 years ago
69e9cb0
RTCOutboundRTPStreamStats[1] added.
by hbos
· 8 years ago
02ba211
Move RTCStatsCollector helper functions to anonymous namespace.
by hbos
· 8 years ago
5d79a7c
rtcstats_objects.h updated with TODOs about stats not being collected
by hbos
· 8 years ago
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